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   32 #define BITSTREAM_READER_LE 
   43 #define MAX_SUBFRAME_COUNT   5 
   73         .bits_per_frame     = 160,
 
   75         .frames_per_packet  = 1,
 
   76         .pitch_sharp_factor = 0.00,
 
   78         .number_of_fc_indexes = 10,
 
   79         .ma_predictor_bits    = 1,
 
   80         .vq_indexes_bits      = {7, 8, 7, 7, 7},
 
   81         .pitch_delay_bits     = {9, 6},
 
   83         .fc_index_bits        = {4, 5, 4, 5, 4, 5, 4, 5, 4, 5},
 
   89         .bits_per_frame     = 152,
 
   91         .frames_per_packet  = 1,
 
   92         .pitch_sharp_factor = 0.8,
 
   94         .number_of_fc_indexes = 3,
 
   95         .ma_predictor_bits    = 0,
 
   96         .vq_indexes_bits      = {6, 7, 7, 7, 5},
 
   97         .pitch_delay_bits     = {8, 5, 5},
 
   99         .fc_index_bits        = {9, 9, 9},
 
  105         .bits_per_frame     = 232,
 
  107         .frames_per_packet  = 2,
 
  108         .pitch_sharp_factor = 0.8,
 
  110         .number_of_fc_indexes = 3,
 
  111         .ma_predictor_bits    = 0,
 
  112         .vq_indexes_bits      = {6, 7, 7, 7, 5},
 
  113         .pitch_delay_bits     = {8, 5, 5},
 
  115         .fc_index_bits        = {5, 5, 5},
 
  121         .bits_per_frame     = 296,
 
  123         .frames_per_packet  = 2,
 
  124         .pitch_sharp_factor = 0.85,
 
  126         .number_of_fc_indexes = 1,
 
  127         .ma_predictor_bits    = 0,
 
  128         .vq_indexes_bits      = {6, 7, 7, 7, 5},
 
  129         .pitch_delay_bits     = {8, 5, 8, 5, 5},
 
  131         .fc_index_bits        = {10},
 
  137     1.0/(1 <<  1), 1.0/(1 <<  2), 1.0/(1 <<  3), 1.0/(1 <<  4),
 
  138     1.0/(1 <<  5), 1.0/(1 <<  6), 1.0/(1 <<  7), 1.0/(1 <<  8),
 
  139     1.0/(1 <<  9), 1.0/(1 << 10), 1.0/(1 << 11), 1.0/(1 << 12),
 
  140     1.0/(1 << 13), 1.0/(1 << 14), 1.0/(1 << 15), 1.0/(1 << 16)
 
  143 static void dequant(
float *
out, 
const int *idx, 
const float * 
const cbs[])
 
  149     for (
i = 0; 
i < num_vec; 
i++)
 
  163         lsfnew[
i] = lsf_history[
i] * 0.33 + lsf_tmp[
i] + 
mean_lsf[
i];
 
  175         lsfnew[
i] = cos(lsfnew[
i]);
 
  186         fixed_vector[
i] += beta * fixed_vector[
i - pitch_lag_int];
 
  202     for (
i = 0; 
i < 5; 
i++)
 
  222     float t, 
t0 = 1.0 / num_subfr;
 
  225     for (
i = 0; 
i < num_subfr; 
i++) {
 
  227             lsfint[j] = lsfold[j] * (1 - t) + t * lsfnew[j];
 
  238 static void eval_ir(
const float *Az, 
int pitch_lag, 
float *freq,
 
  239                     float pitch_sharp_factor)
 
  249     memset(tmp1 + 11, 0, 37 * 
sizeof(
float));
 
  261                                   const float *shape, 
int length)
 
  265     memset(
out, 0, length*
sizeof(
float));
 
  267         for (j = 
pulses->x[
i]; j < length; j++)
 
  316         for (
i = 0; 
i < 3; 
i++) {
 
  317             fixed_sparse->
x[
i] = 3 * (
pulses[
i] & 0xf) + 
i;
 
  318             fixed_sparse->
y[
i] = 
pulses[
i] & 0x10 ? -1 : 1;
 
  323         for (
i = 0; 
i < 3; 
i++) {
 
  324             fixed_sparse->
x[2*
i    ] = 3 * ((
pulses[
i] >> 4) & 0
xf) + 
i;
 
  325             fixed_sparse->
x[2*
i + 1] = 3 * ( 
pulses[
i]       & 0xf) + 
i;
 
  327             fixed_sparse->
y[2*
i    ] = (
pulses[
i] & 0x100) ? -1.0: 1.0;
 
  329             fixed_sparse->
y[2*
i + 1] =
 
  330                 (fixed_sparse->
x[2*
i + 1] < fixed_sparse->
x[2*
i]) ?
 
  331                 -fixed_sparse->
y[2*
i    ] : fixed_sparse->
y[2*
i];
 
  342             for (
i = 0; 
i < 3; 
i++) {
 
  352             int pulse_subset = (
pulses[0] >> 8) & 1;
 
  354             fixed_sparse->
x[0] = ((
pulses[0] >> 4) & 15) * 3 + pulse_subset;
 
  355             fixed_sparse->
x[1] = ( 
pulses[0]       & 15) * 3 + pulse_subset + 1;
 
  357             fixed_sparse->
y[0] = 
pulses[0] & 0x200 ? -1 : 1;
 
  358             fixed_sparse->
y[1] = -fixed_sparse->
y[0];
 
  376     float *synth = 
ctx->synth_buf + 16; 
 
  390     for (
i = 0; 
i < subframe_count; 
i++) {
 
  394         float pitch_gain, gain_code, avg_energy;
 
  408                             ctx->past_pitch_gain < 0.8);
 
  423                                           avg_energy, 
ctx->energy_history,
 
  430         pitch_gain *= 0.5 * pitch_gain;
 
  431         pitch_gain = 
FFMIN(pitch_gain, 0.4);
 
  433         ctx->gain_mem = 0.7 * 
ctx->gain_mem + 0.3 * pitch_gain;
 
  435         gain_code *= 
ctx->gain_mem;
 
  438             fixed_vector[j] = excitation[j] - gain_code * fixed_vector[j];
 
  458         for (
i = 0; 
i < subframe_count; 
i++) {
 
  474                                              (
const float[2]) {-1.99997   , 1.000000000},
 
  475                                              (
const float[2]) {-1.93307352, 0.935891986},
 
  477                                              ctx->highpass_filt_mem,
 
  497                "Invalid block_align: %d. Mode %s guessed based on bitrate: %"PRId64
"\n",
 
  513     for (
i = 0; 
i < 4; 
i++)
 
  514         ctx->energy_history[
i] = -14;
 
  524                              int *got_frame_ptr, 
AVPacket *avpkt)
 
  527     const uint8_t *buf=avpkt->
data;
 
  537                "Error processing packet: packet size (%d) too small\n",
 
  
#define AV_LOG_WARNING
Something somehow does not look correct.
 
static void eval_ir(const float *Az, int pitch_lag, float *freq, float pitch_sharp_factor)
Evaluate the adaptive impulse response.
 
float ff_amr_set_fixed_gain(float fixed_gain_factor, float fixed_mean_energy, float *prediction_error, float energy_mean, const float *pred_table)
Calculate fixed gain (part of section 6.1.3 of AMR spec)
 
void ff_decode_pitch_lag(int *lag_int, int *lag_frac, int pitch_index, const int prev_lag_int, const int subframe, int third_as_first, int resolution)
Decode the adaptive codebook index to the integer and fractional parts of the pitch lag for one subfr...
 
static void postfilter_5k0(SiprContext *ctx, const float *lpc, float *samples)
Apply postfilter, very similar to AMR one.
 
void ff_amrwb_lsp2lpc(const double *lsp, float *lp, int lp_order)
LSP to LP conversion (5.2.4 of AMR-WB)
 
const float ff_b60_sinc[61]
b60 hamming windowed sinc function coefficients
 
This structure describes decoded (raw) audio or video data.
 
void ff_acelp_apply_order_2_transfer_function(float *out, const float *in, const float zero_coeffs[2], const float pole_coeffs[2], float gain, float mem[2], int n)
Apply an order 2 rational transfer function in-place.
 
void ff_sipr_decode_frame_16k(SiprContext *ctx, SiprParameters *params, float *out_data)
 
void ff_sort_nearly_sorted_floats(float *vals, int len)
Sort values in ascending order.
 
#define LP_FILTER_ORDER
linear predictive coding filter order
 
#define MAX_SUBFRAME_COUNT
 
void ff_celp_lp_synthesis_filterf(float *out, const float *filter_coeffs, const float *in, int buffer_length, int filter_length)
LP synthesis filter.
 
const FFCodec ff_sipr_decoder
 
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
 
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
 
#define LSFQ_DIFF_MIN
minimum LSF distance (3.2.4) 0.0391 in Q13
 
uint8_t frames_per_packet
 
#define SUBFR_SIZE
Subframe size for all modes except 16k.
 
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
 
AVCodec p
The public AVCodec.
 
const float ff_pow_0_55[10]
Table of pow(0.55,n)
 
AVChannelLayout ch_layout
Audio channel layout.
 
static void lsf_decode_fp(float *lsfnew, float *lsf_history, const SiprParameters *parm)
 
static void convolute_with_sparse(float *out, const AMRFixed *pulses, const float *shape, int length)
Evaluate the convolution of a vector with a sparse vector.
 
uint8_t number_of_fc_indexes
 
static double val(void *priv, double ch)
 
int gc_index[5]
fixed-codebook gain indexes
 
void ff_adaptive_gain_control(float *out, const float *in, float speech_energ, int size, float alpha, float *gain_mem)
Adaptive gain control (as used in AMR postfiltering)
 
static void decode_parameters(SiprParameters *parms, GetBitContext *pgb, const SiprModeParam *p)
Extract decoding parameters from the input bitstream.
 
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
 
#define FF_CODEC_DECODE_CB(func)
 
static void dequant(float *out, const int *idx, const float *const cbs[])
 
Sparse representation for the algebraic codebook (fixed) vector.
 
int(* init)(AVBSFContext *ctx)
 
static void sipr_decode_lp(float *lsfnew, const float *lsfold, float *Az, int num_subfr)
 
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
 
#define CODEC_LONG_NAME(str)
 
static void decode_fixed_sparse(AMRFixed *fixed_sparse, const int16_t *pulses, SiprMode mode, int low_gain)
 
int ma_pred_switch
switched moving average predictor
 
#define L_INTERPOL
Number of past samples needed for excitation interpolation.
 
uint8_t vq_indexes_bits[5]
size in bits of the i-th stage vector of quantizer
 
int64_t bit_rate
the average bitrate
 
int16_t fc_indexes[5][10]
fixed-codebook indexes
 
int pitch_delay[5]
pitch delay
 
#define AV_CODEC_CAP_CHANNEL_CONF
Codec should fill in channel configuration and samplerate instead of container.
 
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
 
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
 
An AVChannelLayout holds information about the channel layout of audio data.
 
enum AVSampleFormat sample_fmt
audio sample format
 
const float ff_pow_0_7[10]
Table of pow(0.7,n)
 
static int sipr_decode_frame(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, AVPacket *avpkt)
 
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf offset
 
static const float *const lsf_codebooks[]
 
void ff_tilt_compensation(float *mem, float tilt, float *samples, int size)
Apply tilt compensation filter, 1 - tilt * z-1.
 
static void pitch_sharpening(int pitch_lag_int, float beta, float *fixed_vector)
Apply pitch lag to the fixed vector (AMR section 6.1.2).
 
int nb_samples
number of audio samples (per channel) described by this frame
 
#define i(width, name, range_min, range_max)
 
void ff_sipr_init_16k(SiprContext *ctx)
 
static const SiprModeParam modes[MODE_COUNT]
 
static void decode_frame(SiprContext *ctx, SiprParameters *params, float *out_data)
 
#define xf(width, name, var, range_min, range_max, subs,...)
 
uint8_t gc_index_bits
size in bits of the gain codebook indexes
 
const float ff_pow_0_75[10]
Table of pow(0.75,n)
 
void ff_weighted_vector_sumf(float *out, const float *in_a, const float *in_b, float weight_coeff_a, float weight_coeff_b, int length)
float implementation of weighted sum of two vectors.
 
const char * name
Name of the codec implementation.
 
static const float gain_cb[128][2]
 
static const float pred[4]
 
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs.
 
void ff_celp_lp_zero_synthesis_filterf(float *out, const float *filter_coeffs, const float *in, int buffer_length, int filter_length)
LP zero synthesis filter.
 
uint8_t pitch_delay_bits[5]
size in bits of the adaptive-codebook index for every subframe
 
main external API structure.
 
void av_channel_layout_uninit(AVChannelLayout *channel_layout)
Free any allocated data in the channel layout and reset the channel count to 0.
 
static const float mean_lsf[10]
 
Filter the word “frame” indicates either a video frame or a group of audio samples
 
#define SUBFRAME_COUNT_16k
 
float avpriv_scalarproduct_float_c(const float *v1, const float *v2, int len)
Return the scalar product of two vectors.
 
#define AV_CHANNEL_LAYOUT_MONO
 
uint8_t fc_index_bits[10]
size in bits of the fixed codebook indexes
 
This structure stores compressed data.
 
static av_cold int sipr_decoder_init(AVCodecContext *avctx)
 
int gp_index[5]
adaptive-codebook gain indexes
 
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
 
static const int8_t pulses[4]
Number of non-zero pulses in the MP-MLQ excitation.
 
void ff_acelp_interpolatef(float *out, const float *in, const float *filter_coeffs, int precision, int frac_pos, int filter_length, int length)
Floating point version of ff_acelp_interpolate()
 
uint8_t ma_predictor_bits
size in bits of the switched MA predictor
 
void ff_set_min_dist_lsf(float *lsf, double min_spacing, int size)
Adjust the quantized LSFs so they are increasing and not too close.